1. Field of the Invention
The present invention relates to audio coding devices, and more particularly to an audio coding device that encodes audio signals to reduce the data size.
2. Description of the Related Art
Digital audio processing technology and its applications have become familiar to us since they are widely used today in various consumer products such as mobile communications devices and compact disc (CD) players. Digital audio signals are usually compressed with an enhanced coding algorithm for the purpose of efficient delivery and storage. Such audio compression algorithms are standardized as, for example, the Moving Picture Expert Group (MPEG) specifications.
Typical MPEG audio compression algorithms include MPEG1-Audio layer3 (MP3) and MPEG2-AAC (Advanced Audio Codec). MP3 is the layer-3 compression algorithm of the MPEG-1 audio standard, which is targeted to coding of monaural signals or two-channel stereo signals. MPEG-1 Audio is divided into three categories called “layers,” the layer 3 being superior to the other layers (layer 1 and layer 2) in terms of sound qualities and data compression ratios that they provide. MP3 is a popular coding format for distribution of music files over the Internet.
MPEG2-AAC is an audio compression standard for multi-channel signal coding. It has achieved both high audio qualities and high compression ratios while sacrificing compatibility with the existing MPEG-1 audio specifications. Besides being suitable for online distribution of music via mobile phone networks, MPEG2-AAC is a candidate technology for digital television broadcasting via satellite and terrestrial channels. MP3 and MPEG2-AAC algorithms are, however, similar in that both of them are designed to extract frames of a given pulse code modulation (PCM) signal, process them with spatial transform, quantize the resulting transform coefficients, and encode them into a bitstream.
To realize a high-quality coding with maximum data compression, the above MP3 and MPEG2-AAC coding algorithms calculate optimal quantization step sizes (scalefactors), taking into consideration the response of the human auditory system. However, the existing methods for this calculation require a considerable amount of computation. To improve the efficiency of coding without increasing the cost, the development of a new realtime encoder is desired.
One example of existing techniques is found in Japanese Unexamined Patent Publication No. 2000-347679, paragraph Nos. 0059 to 0085 and FIG. 1. According to the proposed audio coding technique, scheduling coefficients and quantization step sizes are changed until the amount of coded data falls within a specified limit while the resulting quantization distortion is acceptable. Another example is the technique disclosed in Japanese Unexamined Patent Publication No. 2000-347679. While attempting to reduce computational loads of audio coding, the disclosed technique takes an iterative approach, as in the above-mentioned existing technique, to achieve a desired code size. Because of a fair amount of time that it spends to reach the convergence of calculation, this technique is not the best for reduction of computational load.